similar to: SIP request send me 482 error

Hi, I was switching from zaptel to dahdi and got latest SVN from everything. Compiling works fine. kernel module dahdi_dummy is loaded. /dev/dahdi/pseudo exists Trying to go into a meetme does not work: [Aug 11 14:04:45] -- Executing [8001 at client_int_sgmobile:1] MeetMe("SCCP/6000-00000001", "444|dcIM") in new stack [Aug 11 14:04:45] WARNING[4184]: app_meetme.c:775

Hi, I have the following symptoms: Call X-lite / Nokia E51 X-lite press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH Call X-lite / SCCP phone MOH works as supposed Call SCCP phone / Nokia E51 SCCP press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH In addition, the BLF on the SCCP phones does NOT show the hinted SIP extension on hold. With 1.4

Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk 1.2.26. I have uploaded in my tftp server the firmware 'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in SEPmacaddress.cnf.xml I have: <loadInformation>SIP11.8-0-4SR1S</loadInformation> ..but in tftp log server I have: Oct 07 11:56:22 asterisk1.local

The Asterisk.org development team is proud to announce the releases of Asterisk 1.4.22 and 1.6.0. ================================================================= === Asterisk 1.4.22 ============================================= ================================================================= Asterisk 1.4.22 includes a large number of bug fixes for the 1.4 release series of Asterisk. 1.4.22

The Asterisk.org development team is proud to announce the releases of Asterisk 1.4.22 and 1.6.0. ================================================================= === Asterisk 1.4.22 ============================================= ================================================================= Asterisk 1.4.22 includes a large number of bug fixes for the 1.4 release series of Asterisk. 1.4.22

>> I've tried specifying an output file as well, per the script's command line options, >> but the output file is 0 bytes.? Does anyone have any suggestions?? I *think* I'm >> using it the way it's intended to be used, but maybe I'm not?! >> -Dave > > Hi! > Maybe the key you tried was not used to encrypt the file? > Aki Aki,

Good morning everybody. My question is simple. Is there a way to perform relay register with Asterisk ? More precisely, I want my clients regiter to a Proxy Registrar (OpenSIPS/Kamailio) through my Asterisk : REGISTER REGISTER Client ------------> Asterisk ---------------> OpenSIPS So Asterisk keep a list of registered clients and only allows them to

Hi, After using a little bit oVirt, here's my thoughts: On the Network part : * In order to start the bridges, you have to reboot the node. It should be good to create the bridges "on the fly". * The VLAN configuration seems not to work (you can create it but it's impossible to assign it to an interface...). You can also assign this kind of interface on a opennet On the VM part:

Hi, I finally managed to have one node working. I added a iSCSI LUN (ok if selinux is set to 0) and wanted to create my first VM. So I created a VM and had some issues: * I can choose boot from HD but I'm just able to choose the LUN, not a specific size in the LUN like you can do in vmware ESX. So, do I have to create a LUN per VM? If yes, it's quite complicated... * If I choose boot from

https://bugzilla.mindrot.org/show_bug.cgi?id=1581 Summary: Pb with syslog Product: Portable OpenSSH Version: 5.2p1 Platform: Sparc OS/Version: Solaris Status: NEW Severity: major Priority: P1 Component: sshd AssignedTo: unassigned-bugs at mindrot.org ReportedBy: eric.savidan at

Add of a button destroy for disabled hosts. This button behave in a similar way than the delete button of a VM. Signed-off-by: Sylvain Desbureaux <sylvain.desbureaux at orange-ftgroup.com> --- src/app/controllers/host_controller.rb | 5 +++++ src/app/services/host_service.rb | 15 +++++++++++++++ src/app/views/host/show.rhtml | 17 +++++++++++++++++ 3 files changed,

Hi Community! If this issue was already topic, please excuse or delete my request... Topic 1 "no ringtone": I configured a SIP registration with my SIP provider (SIPCall). Everything works fine except the ring tone for the caller. The caller hears silence until the called party takes up the phone. I used the DIAL command with the r and R option but no luck... :( Has anybody the same

Hi all, I'm a totally noob on that point but on my test machine, I've made some changes: * Changes to make the ovirt node able to load igb * Changes to make the possibility to destroy an host (if it's disabled) * Changes to make work again (at least on the GUI part) the migration I think I'll should send "somewhere" those patches but : * How do I make the patches (with

Moro, On 5/14/19 4:36 PM, Oskar Vilkevuori wrote: > Hi there, > > Any idea? Please use these packages  https://wiki.xiph.org/Icecast_Server/Installing_latest_version_(official_Xiph_repositories) or rebuild the Debian package but with the openssl -dev package present on your machine. > > I found > from https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=744815 that >

Hi all, I've got some problems to make work oVirt. I've installed a Fedora 10 VM the lightest possible (nothing checked, even based) and installed after some packages (wget, sudo, acpid,...) and updated the system. By the way, acpid should be a dependency of ovirt-installer because the installation fails if it's not available. I've then installed ovirt (ovirt-server-installer

Hi All, I'm running CentOS 5.3 x86, and recently my yum update isn't running as it should. It pops up with sub conflict messages for i386 packages, which I don't need - even though yum is downloading them. Due to this sub conflict messages, I'm not able to update/install my x86 packages, as the installation stops at the error message. Anybody, any idea how to come over

Hello everyone, I have a problem with a FastAGI connection, could you help me fix this problem please? Here is my log: [2015-12-15 16:17:09] WARNING[23936][C-00000015]: res_agi.c:1658 handle_connection: Connecting to '10.171.54.149:9110' failed for url

Hi guys, I'm trying to use Asterisk with LDAP integration. I created some schemas and it seems to work fine for sip.conf replacement. When I try to register a softphone to test the service, it seems ok from the softphone point of view (user registred) but when I do a "sip show peers", no one is registered (nor sip show subrscriptions, users...) I put my Asterisk on full debug and I

That does not make too much sense to me... Configuration should be ok... [Aug 8 23:30:13] SSL certificate ok [Aug 8 23:30:13] == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) [Aug 8 23:30:13] WARNING[23835]: tcptls.c:463 ast_make_file_from_fd: FILE * open failed! Terve, Stefan -- Last words of a stormchaser: "Where is that rotation on the radar?!"

Hi, when I still had ISDN, I was using Hangup(causecode) to send e.g. "Wrong number" to unwelcome callers. Meanwhile, I am only using SIP providers (no PSTN lines any more) and I would like to do similar, i.e. send specific SIP headers. Besides "wrong number", I would especially like to send 302 temp moved with a specified address to deflect certain calls. Is there any way to